With the development of Internet, more and more users communicate with each other on the Internet. With users' further requirements for communication and the further development of the Internet, the users can communicate with each other on the Internet not only through characters but also through video-audio data.
In the prior art, since data transmission on the Internet is based on a Transmission Control Protocol/Internet Protocol (TCP/IP), when video-audio data are transmitted on the Internet, it is needed to divide the video-audio data into multiple data packets according to the TCP/IP and transmit the multiple data packets on the Internet by taking a frame as a unit. Since the network structure of the Internet is complex, the transmission rate of the data packets transmitted on the Internet can not be controlled effectively, and a receiving end of the video-audio data sometimes can not receive the video-audio data for a long time, which results in disconnection or incontinuity phenomena when the video-audio data is recovered and played, e.g. audio incontinuity, call mute, video image standstill and so on.
In order to solve the problem, a buffer is usually configured on the receiving end of the video-audio data to store some video-audio data, and the video-audio data are received from the network, and then decoded and to be played, so as to decrease the above phenomena. But the capacity of the buffer is limited, if the buffer does not receive subsequent video-audio data after the video-audio data in the buffer has been played, the incontinuity phenomena will occur when video-audio data is played, which decreases user experiences.